webrtc/sdk/android
Taylor Brandstetter cdd05f0cc1 Implement proper SCTP data channel closing procedure.
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!

https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7

However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.

This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.

This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.

TBR=magjed@webrtc.org

Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
2018-05-31 21:01:53 +00:00
..
api/org/webrtc Add min_bitrate_bps to RtpEncodingParameters. 2018-05-31 10:17:57 +00:00
instrumentationtests Implement proper SCTP data channel closing procedure. 2018-05-31 21:01:53 +00:00
native_api Android JNI generation: Set JNI namespace in build files 2018-05-18 10:44:38 +00:00
native_unittests Android JNI generation: Set JNI namespace in build files 2018-05-18 10:44:38 +00:00
src Add min_bitrate_bps to RtpEncodingParameters. 2018-05-31 10:17:57 +00:00
tests/src/org/webrtc Add VideoEncoder.ScalingSettings.toString method. 2018-04-18 14:36:19 +00:00
AndroidManifest.xml Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
BUILD.gn Reland "Injectable logging" 2018-05-18 15:04:16 +00:00
OWNERS Android: Add henrika@ as owner of audio code 2018-03-21 09:59:18 +00:00
PRESUBMIT.py Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README Updating android/README. 2018-03-01 20:22:48 +00:00

This directory holds a Java implementation of the webrtc::PeerConnection API, as
well as the JNI glue C++ code that lets the Java implementation reuse the C++
implementation of the same API.

To build the Java API and related tests, make sure you have a WebRTC checkout
with Android specific parts. This can be used for linux development as well by
configuring gn appropriately, as it is a superset of the webrtc checkout:
fetch --nohooks webrtc_android
gclient sync

You also must generate GN projects with:
--args='target_os="android" target_cpu="arm"'

More information on getting the code, compiling and running the AppRTCMobile
app can be found at:
https://webrtc.org/native-code/android/

To use the Java API, start by looking at the public interface of
org.webrtc.PeerConnection{,Factory} and the org.webrtc.PeerConnectionTest.

To understand the implementation of the API, see the native code in src/jni/pc/.